1. Field of the Invention
The present invention relates to a communication terminal and, more particularly, to a systems and methods of generating and transmitting a ring back tone.
2. Description of the Related Art
Generally, if a calling party dials to make a call using a wire or wireless phone, the calling party hears a ring tone for informing that a calling signal is being sent to a called party. This tone is, typically, referred to as a ring back tone.
When a call connection is attempted between communication terminals of a public switched telephone network (PSTN), a switching system of the terminal of the called party generates the ring back tone and transmits the ring back tone to the terminal of the calling party.
Referring to FIG. 1, a communication system using the conventional Internet Protocol phone (IP phone) is illustrated in a general communication network. A call connection method according to the E.164 standard of the PSTN is based on a method of connecting a calling party and a called party together using the phone number of the called party.
That is, if the calling party inputs a phone number of a specified called party, the PSTN establishes a communication path connecting the calling party and the specified called party using the phone number inputted by the calling party.
Distinct from a communication network constructed for the main purpose of voice communication such as the PSTN, a data communication network constructed for the purpose of data communication is classified into a local area network (LAN), wide area network (WAN), and the Internet, in accordance with its size. Most data communication networks transmit/receive data in the form of a packet in accordance with the characteristics of the data communication protocols implemented.
Voice may be also transmitted using a data communication network such as an Internet protocol (IP) network. Voice transmission using a commercialized packet data network is called a phone-to-phone service. For example, if a calling party ‘a’ connects to a data communication network of a neighboring IP telephone company ‘A’ through the existing PSTN and inputs the phone number of a called party ‘b’, the calling party ‘a’ is connected to a data communication network of another IP telephone company ‘B’ near the called party ‘b’ through an IP network, and the data communication network of IP telephone company ‘B’ connects the calling party to the called party ‘b’ through another PSTN.
Currently, a Personal Computer (PC)-to-PSTN phone and an IP phone-to-IP phone have been proposed. For further detail please refer to H.323 recommendation as the international standard for voice communication of PC-to-PC, PC-to-PSTN, and IP phone-to-IP phone.
The H.323 recommendation is the standard of the International Telecommunications Union—Telecommunication Standardization Sector (ITU-T) for transmitting multimedia video-conference data through a packet exchange type network such as a transmission control protocol/Internet protocol (TCP/IP).
Referring back to FIG. 1, if a calling party of a PSTN phone makes a phone call to a called party of an IP phone, the PSTN phone is connected to an IP network through a trunk gateway, and the trunk gateway is connected to the IP phone through a soft switch or an IP phone server in the IP network. That is, the connection between PSTN phone and IP phone is made through the path illustrated graphically as {circle around (1)}→{circle around (2)}→{circle around (3)}→{circle around (4)}→{circle around (5)}→{circle around (6)}→{circle around (7)}→{circle around (8)}→{circle around (9)}→{circle around (10)} in FIG. 1.
The IP phone server includes a gatekeeper, a proxy server, and a call controller. The gatekeeper, in accordance with the H.323 recommendation, takes charge of E.164/IP address translation, admission control, bandwidth control, call control, use of call routing/control resources, security function, etc.
The proxy server serves to store therein a call management request message received from a user agent (UA) such as call setup, call cancellation, call termination, etc., in a voice over IP (VoIP).
The proxy server simultaneously sends the call management request message to various addresses of session initiation protocol (SIP) registered and therefore the same user. Also, if a user agent's response to the call management request is received by the proxy server, the proxy server serves to transmit the best response to the user agent (UA), and to process the cancellation of other messages generated simultaneously.
The gateway connects a data communication network with a switched network such as the PSTN. Also, the gateway provides bandwidth/medium control and also protocol/medium conversion function. The gateway is a network point and serves as an entrance to another network, and may be classified into three kinds. That is, a trunk gateway for connecting switching systems, an access gateway for directly connecting terminals, and a residential gateway for home use.
A soft switch in FIG. 1 is the platform that serves as a bridge for other kinds of signaling systems, and controls various types of media gates. The soft switch is software that serves as a switching system in the existing packet switching network as an upper layer of the gatekeeper in the H.323 recommendation, and takes charge of an integrated private exchange on the Internet capable of integrally managing communication information such as voice, data, image, etc.
To provide a ring back tone toward the PSTN phone in case of making a call to an IP phone from the PSTN phone, the IP phone server or a soft switch connected to the IP phone should generate the ring back tone, but none of the IP phone server and the soft switch have a module for generating the ring back tone. As a result, the ring back tone cannot be transmitted to the PSNT phone.
Conventionally, in case of attempting call connection between IP phones, a calling party of IP phone mainly generates and provides a ring back tone to a user of the calling party, and in case of attempting the call connection from an IP phone to a PSTN phone, a gateway transmits a ring back tone to the IP phone.
Unfortunately, however, a trunk gateway for connecting the PSTN phone and the IP phone together does not have a module for generating the ring back tone. As such, the ring back tone cannot be sent to the PSTN phone from the IP phone.
Typically, in case of attempting the call connection between the PSTN phones, a mobile switching center of a called party generates and transmits the ring back tone. In case of connection between the IP phones, an access gateway generates and transmits the ring back tone to a calling party of the IP phone.
Otherwise, a called party of the IP phone itself generates the ring back tone and provides a calling party of the IP phone. Distinct from an access gateway connected to a terminal, a trunk gateway for connecting between the switching systems does not include a device for generating a ring back tone.
Unfortunately, in the current systems, when a call is made from a PSTN phone to an IP phone, a gateway between the PSTN and the packet data communication network may not generate the ring back tone. That is, since a trunk gateway for connecting between switching systems does not include a device for generating the ring back tone, a calling party of the PSTN phone may not hear any sound (i.e., calling signal). In other words, the calling party of the PSTN phone may not recognize whether the call attempt is in progress until a called party of the IP phone picks up the phone.
Consequently, when a call is made from an IP phone to a PSTN phone, a called party of the IP phone generates the ring back tone for itself. But when a call is made from a PSTN phone to an IP phone, a trunk gateway may not generate and transmit the ring back tone to a calling party of the PSTN phone.
Methods and systems are needed to provide a solution to the above-mentioned problems.